Commit Graph

7142 Commits

Author SHA1 Message Date
stefan@webrtc.org
d839e0ab52 Reduce to 2 probes when probing for initial bandwidth.
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23359005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7621 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 19:33:55 +00:00
stefan@webrtc.org
db26247a9b Add UMA for measuring the diff between the BWE at 2 seconds compared to the BWE at 20 seconds when the BWE should have converged.
BUG=crbug/425925
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30819005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7620 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 19:32:10 +00:00
mcasas@webrtc.org
8944c9d08b AppRTCDemoActivity: use differnet Themes for different API levels
The current AndroidManifest.xml hardcodes a Theme that
is only available in Android L or later (Material). To
maintain backwards compatibility, and for better App
style, create a single Theme/Style and define it for
different APIs.

I tested this in two Nexus %, one with prerelease L
and another with a KK 4.4.2 and the Theme is indeed
automagically updated :)

Note that this is compatible with
https://webrtc-codereview.appspot.com/26979004/

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7619 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 17:26:22 +00:00
kjellander@webrtc.org
d367321a3f Add kjellander as PRESUBMIT.py OWNER
BUG=
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31949005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7618 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 17:06:31 +00:00
sprang@webrtc.org
dcebf2daa7 Reworked paced sender queue
Packet queue in the paced sender is now based on a priority queue rather than having a separate fifo-queue per priority level. This allows more flexible sorting and cleaner usage.

Packets with earlier capture times are now prioritized higher. In situations with high packet loss, the queue might contain packets from several subsequent frames. Retransmit packets from the earlier frames first, since the later ones will probably be dependent on these.

Also, don't force sending of packets after a certain time of inactivity or when packets grow too old, since this was causing consistent overuse on poor connections. Instead, drop frames in vie encoder if pacer queue is too long.

BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7617 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 16:27:16 +00:00
pbos@webrtc.org
fad9aecbf5 Remove protected files from talk/PRESUBMIT.py.
All files may now be committed to.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/23359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7616 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 16:06:35 +00:00
pbos@webrtc.org
88ef632286 Falling back on single-stream on multiple SSRC.
Instead of failing, use one stream. Also clamp video min bitrate.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/31949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7615 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 15:29:29 +00:00
henrike@webrtc.org
28af64105b Presubmit was not whitelisting libjingle_tests.gyp or sound.gyp due to a missing comma leading to a concatenation of the two strings in the whitelist.
R=kjellander@webrtc.org
TBR=niklas.enbom@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/25999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7614 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 15:11:46 +00:00
stefan@webrtc.org
b3265accd9 Adds support for finch experiments to video_loopback.
Adds support for logging to stderr via -logs.

Enables abs-send-time by default.

R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7613 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 14:57:14 +00:00
henrik.lundin@webrtc.org
52b42cb069 Fix problem with late packets in NetEq
Since r7255, it could happen that an old packet would block the decoding
process until enough packet was received for the buffer to flush. This
CL fixes that by:
- Partially reverting r7255;
- Remove recent old packets before taking a decision for GetAudio;
- Remove all old packets after a packet has been extracted for decoding;
- Adding tests for reordered packets.

BUG=chrome:423985
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7612 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 14:03:58 +00:00
pbos@webrtc.org
09cc686c8b Delete VideoReceiveStream channels in destructor.
R=stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/31909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 13:48:15 +00:00
kwiberg@webrtc.org
6de75ca3ed Remove the useless dummy state parameter to WebRtcPcm16b_DecodeW16
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7610 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 13:29:24 +00:00
kwiberg@webrtc.org
c78cf97ecb Remove the useless dummy state parameter to WebRtcG711_*
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7609 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 13:23:36 +00:00
perkj@webrtc.org
b5d045e94d ReAdd PeerConnectionInterface::AddStream to fix Chrome build.
AddStream(MediaStreamInterface* stream, const MediaConstraintsInterface* constraints);
This will be removed once Chrome has been updated.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7608 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 13:01:33 +00:00
tommi@webrtc.org
18de6f9622 Change the PeerConnection proxy templates to use blocking method calls instead of using Thread::Send.
The problem with Thread::Send is that it processes incoming pending messages and for the proxies,
this can mean that multiple incoming calls can concurrently run on the same thread, resulting in unexpected behavior.

See e.g. crbug.com/429740 (and more)

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7607 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 12:08:48 +00:00
kwiberg@webrtc.org
721ef633d0 Remove the codec_type_ member from AudioDecoder
It isn't actually required, as evidenced by the comparative ease with
which it can be removed.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7606 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 11:51:46 +00:00
perkj@webrtc.org
c2dd5ee2c0 Prepare for removal of PeerConnectionObserver::OnError.
Prepare for removal of constraints to PeerConnection::AddStream.

OnError has never been implemented and has been removed from the spec.
Also, constraints to PeerConnection::AddStream has also been removed from the spec and have never been implemented.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7605 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 11:31:29 +00:00
stefan@webrtc.org
f37145f685 Enables AIMD control by default.
BUG=1788
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7604 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 09:08:21 +00:00
henrik.lundin@webrtc.org
b0f4b3da05 Improving error message from neteq_rtpplay
If a packet with unknown RTP payload type is inserted, this CL
will make sure that the error message is a little more detailed
and gives a better understadning of what to do.

BUG=2692
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7603 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 08:53:10 +00:00
buildbot@webrtc.org
a663d90ae3 (Auto)update libjingle 79104430-> 79104922
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7602 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 22:29:18 +00:00
glaznev@webrtc.org
5f38c8d1b8 Android AppRTCDemo improvements:
- Add a room list to ConnectActivity with buttons to add/remove rooms.
- Add loopback call button.
- Add option to toggle full screen / letterbox video.
- Add camera fps settings.
- Fix device to landscape orientation for HD video until issue 3936
will be fixed.
- Fix a few crashes by avoiding calling peer connection and
GAE signaling function while connection is closing.
- Better handling GAE channel error - catch channel exceptions and
display dialog with error messages.

BUG=3939, 3935
R=kjellander@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7601 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 22:18:52 +00:00
andrew@webrtc.org
5804936052 Add format members to AudioConverter for DCHECKing.
And use a std::min. Post-commit fixes after:
https://review.webrtc.org/30779004/

TBR=kwiberg

Review URL: https://webrtc-codereview.appspot.com/25059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7600 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 21:32:14 +00:00
marpan@webrtc.org
e451b756a8 Update rate control parameter in vp9 test.
TBR=hellner@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7599 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 21:26:08 +00:00
marpan@webrtc.org
4765ca55f9 Roll chromium_revision: 28d1981..d3db2ff
Pick up the libvpx roll: https://codereview.chromium.org/674753002

Summary of changes (28d1981..d3db2ff/DEPS):
* third_party/android_tools 36bf7ac..ea50ccc
* third_party/boringssl 7ea8481..751e889
* third_party/icu 8ac906f..d8b2a9d
* third_party/libvpx efe9712..2e5ced5
* third_party/usrsctp/usrsctplib
* tools/gyp 1990:1991
* tools/swarming_client a57d7db..bcb3bc3

Clang is not updated in this roll.

Made the change getchar() --> getc(stdin) as seems like getchar() isn't supported on android anymore.
(getchar() was causing the error: undefined reference to '__srget')

Update rate control parameter in vp9 test.

R=andrew@webrtc.org
TBR=ajm@google.com

Review URL: https://webrtc-codereview.appspot.com/23229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7598 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 20:10:26 +00:00
andrew@webrtc.org
f866b2d9f9 Restore the void return type on WriteWavHeader.
Karl pointed out that the user can check the validity of the input
parameters with CheckWavParameters prior to calling.

TBR=kwiberg

Review URL: https://webrtc-codereview.appspot.com/23339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7597 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 18:20:06 +00:00
andrew@webrtc.org
b81e304ac0 replace inline assembly WebRtcNsx_AnalysisUpdate by intrinsics.
The modification only uses the unique part of the analysis_update
 function. Pass byte to byte conformance test on both ARMv7 and AArch64,
 and the single function performance is similar with original assembly
 version on different platforms. If not specified, the code is compiled
 by GCC 4.6. The result is the "X version / C version" ratio, and the
 less is better.

| run 100k times             | cortex-a7 | cortex-a9 | cortex-a15 |
| use C as the base on each  |  (1.2Ghz) |  (1.0Ghz) |   (1.7Ghz) |
| CPU target                 |           |           |            |
|----------------------------+-----------+-----------+------------|
| Neon asm                   |    15.61% |    20.15% |     14.89% |
| Neon inline asm (LLVM 3.4) |    25.98% |    33.96% |     18.18% |
| Neon intrinsics (GCC 4.6)  |    22.06% |    27.01% |     19.24% |
| Neon intrinsics (GCC 4.8)  |    17.28% |    18.23% |     18.55% |
| Neon intrinsics (LLVM 3.4) |    21.02% |    19.98% |     16.76% |

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28849004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7596 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 17:17:51 +00:00
henrik.lundin@webrtc.org
f9471807a2 Add Opus support to neteq_rtpplay
BUG=2416
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7595 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 15:19:58 +00:00
pbos@webrtc.org
96a93259b3 Implement external decoder support in WebRtcVideoEngine2.
R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/30839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7594 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 14:46:44 +00:00
stefan@webrtc.org
548b228c91 Add UMA metrics for the initial (after two seconds) packet loss, round-trip time and bandwidth estimate of a WebRTC call.
BUG=crbug/425925
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7593 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 14:42:43 +00:00
asapersson@webrtc.org
96dc685143 Add stats for video:
- number of sent/received RTCP NACK/FIR/PLI per minute
- percentage of unique sent/received NACK requests
- percentage of discarded/duplicated packets by the jitter buffer
- permille of sent/received key frames

BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7592 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 14:40:38 +00:00
henrik.lundin@webrtc.org
2236267b5e Disable PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate under MSan
This test is flaky on MSan bots.

BUG=3980
TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7591 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 13:38:50 +00:00
braveyao@webrtc.org
bf09976e86 Add more sanity checks to workaround the unidentified problem that CaptureThread is still running while related resouces are destroyed already.
BUG=
TEST=auto test
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7590 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 09:58:40 +00:00
marpan@webrtc.org
ed45896759 Adjust/increase rate control thresold for a vp9 test.
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7589 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 07:08:52 +00:00
marpan@webrtc.org
5b88317820 Add VP9 codec to VCM and vie_auto_test.
Include VP9 tests in videoprocessor_integrationtests.
Include end-to-end send/receiveVP9 test.

This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in:
see https://code.google.com/p/webrtc/issues/detail?id=3932

R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
kjellander@webrtc.org
5072e0f6cd Update Android projects to API level 21.
The update in https://webrtc-codereview.appspot.com/23309004
was not enough, so this updates to 21 instead.

This is required in order to roll chromium_revision to
keep up with Chrome, as third_party/android_tools have now
dropped support for API level 20.

Commands used:
third_party/android_tools/sdk/tools/android update project --name OpenSlDemo --target android-21 --path webrtc/examples/android/opensl_loopback
third_party/android_tools/sdk/tools/android update project --name WebRTCDemo --target android-21 --path webrtc/examples/android/media_demo/
third_party/android_tools/sdk/tools/android update project --name AppRTCDemo --target android-21 --path talk/examples/android/
Then I restored the changes of the ANDROID_SDK_ROOT -> ANDROID_HOME since it seems the Chromium build toolchain doesn't set it properly when
build/android/envsetup.sh is sourced.

BUG=
R=glaznev@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7587 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 23:26:10 +00:00
andrew@webrtc.org
818c9f9e14 replace inline assembly WebRtcNsx_SynthesisUpdateNeon by intrinsics.
The modification only uses the unique part of the synthesis_update
function. Pass byte to byte conformance test both on ARMv7 and ARMv8,
and the single function performance is similar with original assembly
version on different platforms (if not specified, the code is compiled
by GCC 4.6):

| run 100k times             | cortex-a7 | cortex-a9 | cortex-a15 |
| use C as the base          |  (1.2Ghz) |  (1.0Ghz) |   (1.7Ghz) |
| (the smaller the better)   |           |           |            |
|----------------------------+-----------+-----------+------------|
| C                          |      100% |      100% |       100% |
| Neon asm                   |    15.93% |    17.01% |     12.50% |
| Neon inline asm            |    27.74% |    31.41% |     14.64% |
| Neon intrinsics (GCC 4.8)  |    17.84% |    14.10% |     13.84% |
| Neon intrinsics (LLVM 3.4) |    16.63% |    14.01% |     12.98% |

BUG=3580
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23159004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7586 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 22:07:35 +00:00
andrew@webrtc.org
a3ed713dad Add a WavReader counterpart to WavWriter.
Don't bother with a C interface as we currently have no need to call
this from C code. The first use will be in the audioproc tool.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7585 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 21:51:03 +00:00
kjellander@webrtc.org
c2c94a9a9f Change default JVM location to /usr/lib/jvm/java-7-openjdk-amd64
Given that OpenJDK 1.7 is the recommended Java SDK for
Chromium these days, we should get rid of linking to the old
non-standardized link referring to a Sun Java 1.6 SDK.

Instead of requiring all users to set JAVA_HOME, I prefer
have the most common path as default and and close webrtc:2113
as won't fix after this is submitted.

BUG=2113
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7584 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 19:01:41 +00:00
kjellander@webrtc.org
78c222bfae Update all .isolate files for the new format.
R=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27809004

Patch from Marc-Antoine Ruel <maruel@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7583 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 18:08:09 +00:00
kjellander@webrtc.org
8a130c1084 Update Android projects to API level 20.
This is required in order to roll chromium_revision to
keep up with Chrome, as third_party/android_tools have now
dropped support for API level 19.

Commands used:
third_party/android_tools/sdk/tools/android update project --name OpenSlDemo --target android-20 --path webrtc/examples/android/opensl_loopback
third_party/android_tools/sdk/tools/android update project --name WebRTCDemo --target android-20 --path webrtc/examples/android/media_demo/
third_party/android_tools/sdk/tools/android update project --name AppRTCDemo --target android-20 --path talk/examples/android/
Then I restored the changes of the ANDROID_SDK_ROOT -> ANDROID_HOME since it seems the Chromium build toolchain doesn't set it properly when
build/android/envsetup.sh is sourced.

BUG=
R=glaznev@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7582 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 17:13:37 +00:00
glaznev@webrtc.org
053c6abf8d Fix N7 camera aspect ratio.
N7 video preview generates stretched output:
https://code.google.com/p/android/issues/detail?id=70830.
To workaround the problem set camera picture size in
addition to video preview size with the same resolution.

BUG=3971
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7581 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 16:58:58 +00:00
andrew@webrtc.org
508c91683c Build fix for MIPS32R6.
Exclude MIPS optimizations for MIPS32R6 build since some of the instructions
are not supported. This is temporary fix, until the MIPS32R6 code is added.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25989004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7580 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 16:26:17 +00:00
andrew@webrtc.org
cc476aa038 Fix a name collision with Android libc++
The Android libc++ has a symbol called '_P'
This CL renames a property called _P in webrtc.

BUG=chromium:427718
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30009004

Patch from Fabrice de Gans-Riberi <fdegans@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7579 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 16:01:25 +00:00
pbos@webrtc.org
b7ed7799e7 Implement conference-mode temporal-layer screencast.
Renames VideoStream::temporal_layers to temporal_layer_thresholds_bps to
convey that it contains thresholds needed to ramp up between them (1
threshold -> 2 temporal layers, etc.).

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788,1667

Review URL: https://webrtc-codereview.appspot.com/23269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7578 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 13:08:10 +00:00
pbos@webrtc.org
3bf3d238c8 Configure A/V sync in WebRtcVideoEngine2.
Sets up A/V sync for the first video receive channel with the default
voice channel. This is only done when conference mode is disabled to
preserve existing behavior. Ideally we'd know which voice channel to
sync with here.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/23249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7577 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 12:59:34 +00:00
stefan@webrtc.org
4abadab708 Simplify bwe tests.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7576 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 10:47:12 +00:00
minyue@webrtc.org
2dc6f3154d Adapting bitrate according to maxplaybackrate for Opus.
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7575 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 05:33:10 +00:00
andrew@webrtc.org
8328e7c44d Revert "Revert part of r7561, "Refactor audio conversion functions.""
This restores the conversion changes to AudioProcessing originally
added in r7561, with minor alterations to ensure it passes all tests.

TBR=kwiberg

Review URL: https://webrtc-codereview.appspot.com/28899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7574 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 04:58:14 +00:00
tkchin@webrtc.org
14146e40aa arm64 iOS build.
Allows successful build of arm64 libraries using
GYP_DEFINES="OS=ios target_arch=arm64 target_subarch=arm64".
Note that not all libraries will be NEON optimized (eg common_audio),
however most importantly libvpx will be. WEBRTC_ARCH_ARM needs to be
defined so that libvpx doesn't post-process, which is significantly
detrimental to performance.

BUG=3898
R=kjellander@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7573 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 00:14:39 +00:00
jiayl@webrtc.org
50ca986bc1 Improve the logging when a TCP connection is deleted.
BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7572 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30 23:50:54 +00:00